Hello,
There is no difference in the version. The successful test was to show the
differences and the client was registering to an asterix PBX. Both versions
work fine with asterisk, none work with ser.
Regards,
Jorge
On Wednesday 03 December 2003 13:41, Jan Janak wrote:
Hello,
the unsucessfull test was made with kphone 3.12 and the sucessfull one
with 3.14. Try to upgrade the older version to 3.14 as well. Also try to
configure it the same way (i.e. use IP instead of 'asterix' in the
configuration).
Jan.
On 02-12 09:05, Jorge Merlino wrote:
> Nils:
>
> I'm afraid there is no more interesting dump to send you, kphone keeps
> sending the same register message and SER keeps answering the same.
> By the way yesterday I tried to register kphone without authentication
> configured in SER and it didn't work either. Kphone keeps trying again
> and again although it reiceives an OK answer from SER every time.
> Oddly I've tested kphone with other proxies and always worked fine.
> Here is a dump of this other situation (by the way I'm using kphone v.
> 3.12):
>
>
> SipClient: Listening UDP on port: 5060
> SipClient: Our address: 192.168.2.175
> SipRegister: Auth is '(null)'
> SipRegister: Proxy Auth is '(null)'
>
> SipClient: Sending: 09:00:38.025
> --------------------------------
> REGISTER sip:asterix SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.175
> CSeq: 4764 REGISTER
> To: "jorge" <sip:jorge@asterix>
> Expires: 900
> From: "jorge" <sip:jorge@asterix>
> Call-ID: 123630670(a)192.168.2.175
> Content-Length: 0
> User-Agent: KPhone/3.12
> Event: registration
> Allow-Events: presence
> Contact: "jorge"
<sip:jorge@192.168.2.175;transport=udp>;methods="INVITE,
> MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
>
>
> SipClient: Sending to '192.168.2.176:5060'
> SipClient: Receiving message...
>
> SipClient: Received: 09:00:38.041
> ---------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.175
> CSeq: 4764 REGISTER
> To: "jorge"
<sip:jorge@asterix>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22
> From: "jorge" <sip:jorge@asterix>
> Call-ID: 123630670(a)192.168.2.175
> Contact: <sip:jorge@192.168.2.175:5062;transport=udp>;q=0.00;expires=337
> Contact: <sip:jorge@192.168.2.175;transport=udp>;q=0.00;expires=900
> Server: Sip EXpress router (0.8.11 (i386/linux))
> Content-Length: 0
>
>
> SipTransaction: Retransmit 1 (4000)
>
> SipClient: Sending: 09:00:42.026
> --------------------------------
> REGISTER sip:asterix SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.175
> CSeq: 4764 REGISTER
> To: "jorge" <sip:jorge@asterix>
> Expires: 900
> From: "jorge" <sip:jorge@asterix>
> Call-ID: 123630670(a)192.168.2.175
> Content-Length: 0
> User-Agent: KPhone/3.12
> Event: registration
> Allow-Events: presence
> Contact: "jorge"
<sip:jorge@192.168.2.175;transport=udp>;methods="INVITE,
> MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
>
>
> SipClient: Receiving message...
>
> SipClient: Received: 09:00:42.038
> ---------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.175
> CSeq: 4764 REGISTER
> To: "jorge"
<sip:jorge@asterix>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22
> From: "jorge" <sip:jorge@asterix>
> Call-ID: 123630670(a)192.168.2.175
> Contact: <sip:jorge@192.168.2.175:5062;transport=udp>;q=0.00;expires=333
> Contact: <sip:jorge@192.168.2.175;transport=udp>;q=0.00;expires=900
> Server: Sip EXpress router (0.8.11 (i386/linux))
> Content-Length: 0
>
> -------------------------------------------------------------------------
>----------------------------
>
> Here is an example of a successful connection with other proxy (asterisk
> in this case)
>
> SipClient: Listening UDP on port: 5062
> SipClient: Our address: 192.168.2.176
> SipRegister: Auth is '(null)'
> SipRegister: Proxy Auth is '(null)'
>
> SipClient: Sending: 08:47:08.026
> --------------------------------
> REGISTER sip:192.168.2.175 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.176:5062;rport
> CSeq: 1124 REGISTER
> To: "Luis Vazquez" <sip:101@192.168.2.175>
> Expires: 600
> From: "Luis Vazquez" <sip:101@192.168.2.175>
> Call-ID: 546198833(a)192.168.2.176
> Content-Length: 0
> User-Agent: KPhone/3.14
> Event: registration
> Allow-Events: presence
> Contact: "Luis Vazquez"
> <sip:luis@192.168.2.176:5062;transport=udp>;methods="INVITE, MESSAGE,
> INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
>
>
> SipClient: Sending to '192.168.2.175:5060'
> SipClient: Receiving message...
>
> SipClient: Received: 08:47:08.048
> ---------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.2.176:5062;rport
> From: "Luis Vazquez" <sip:101@192.168.2.175>
> To: "Luis Vazquez" <sip:101@192.168.2.175>;tag=as3f7be343
> Call-ID: 546198833(a)192.168.2.176
> CSeq: 1124 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:101@192.168.2.175>
> Content-Length: 0
>
>
> SipCall: Incoming response
> SipTransaction: Incoming Response
> SipClient: Receiving message...
>
> SipClient: Received: 08:47:08.051
> ---------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.176:5062;rport
> From: "Luis Vazquez" <sip:101@192.168.2.175>
> To: "Luis Vazquez" <sip:101@192.168.2.175>;tag=as3f7be343
> Call-ID: 546198833(a)192.168.2.176
> CSeq: 1124 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Expires: 600
> Contact: <sip:101@192.168.2.175>;expires=600
> Date: Tue, 02 Dec 2003 11:46:29 GMT
> Content-Length: 0
>
>
> SipCall: Incoming response
> SipCall: Checking for Contact and Record-Route
> SipCall: Setting Contact for this Call Member
> SipTransaction: Incoming Response
> ReRegistrationTimer (ms): 540000
>
>
> Regards
> Jorge
>
>
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